avfilter: add compensation delay line filter

Signed-off-by: Paul B Mahol <onemda@gmail.com>

6 files changed, 250 insertions, 1 deletions

diff --git a/Changelog b/Changelog
index 380572bda2..f082aa3d56 100644
--- a/Changelog
+++ b/Changelog
@@ -34,6 +34,7 @@ version <next>:
 - realtime filter
 - anoisesrc audio filter source
 - IVR demuxer
+- compensationdelay filter
 
 
 version 2.8:
diff --git a/doc/filters.texi b/doc/filters.texi
index a7f8a53f36..1d03cee0e3 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -1628,6 +1628,54 @@ compand=.1|.1:.1|.1:-45.1/-45.1|-45/-900|0/-900:.01:45:-90:.1
 @end example
 @end itemize
 
+@section compensationdelay
+
+Compensation Delay Line is a metric based delay to compensate differing
+positions of microphones or speakers.
+
+For example, you have recorded guitar with two microphones placed in
+different location. Because the front of sound wave has fixed speed in
+normal conditions, the phasing of microphones can vary and depends on
+their location and interposition. The best sound mix can be achieved when
+these microphones are in phase (synchronized). Note that distance of
+~30 cm between microphones makes one microphone to capture signal in
+antiphase to another microphone. That makes the final mix sounding moody.
+This filter helps to solve phasing problems by adding different delays
+to each microphone track and make them synchronized.
+
+The best result can be reached when you take one track as base and
+synchronize other tracks one by one with it.
+Remember that synchronization/delay tolerance depends on sample rate, too.
+Higher sample rates will give more tolerance.
+
+It accepts the following parameters:
+
+@table @option
+@item mm
+Set millimeters distance. This is compensation distance for fine tuning.
+Default is 0.
+
+@item cm
+Set cm distance. This is compensation distance for tightening distance setup.
+Default is 0.
+
+@item m
+Set meters distance. This is compensation distance for hard distance setup.
+Default is 0.
+
+@item dry
+Set dry amount. Amount of unprocessed (dry) signal.
+Default is 0.
+
+@item wet
+Set wet amount. Amount of processed (wet) signal.
+Default is 1.
+
+@item temp
+Set temperature degree in Celsius. This is the temperature of the environment.
+Default is 20.
+@end table
+
 @section dcshift
 Apply a DC shift to the audio.
 
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 1f4abeb97c..c89637406f 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -64,6 +64,7 @@ OBJS-$(CONFIG_CHANNELMAP_FILTER)             += af_channelmap.o
 OBJS-$(CONFIG_CHANNELSPLIT_FILTER)           += af_channelsplit.o
 OBJS-$(CONFIG_CHORUS_FILTER)                 += af_chorus.o generate_wave_table.o
 OBJS-$(CONFIG_COMPAND_FILTER)                += af_compand.o
+OBJS-$(CONFIG_COMPENSATIONDELAY_FILTER)      += af_compensationdelay.o
 OBJS-$(CONFIG_DCSHIFT_FILTER)                += af_dcshift.o
 OBJS-$(CONFIG_DYNAUDNORM_FILTER)             += af_dynaudnorm.o
 OBJS-$(CONFIG_EARWAX_FILTER)                 += af_earwax.o
diff --git a/libavfilter/af_compensationdelay.c b/libavfilter/af_compensationdelay.c
new file mode 100644
index 0000000000..33ee7e4996
--- /dev/null
+++ b/libavfilter/af_compensationdelay.c
@@ -0,0 +1,198 @@
+/*
+ * Copyright (c) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen, Vladimir Sadovnikov and others
+ * Copyright (c) 2015 Paul B Mahol
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "libavutil/opt.h"
+#include "libavutil/samplefmt.h"
+#include "avfilter.h"
+#include "audio.h"
+#include "internal.h"
+
+typedef struct CompensationDelayContext {
+    const AVClass *class;
+    int distance_mm;
+    int distance_cm;
+    int distance_m;
+    double dry, wet;
+    int temp;
+
+    unsigned delay;
+    unsigned w_ptr;
+    unsigned buf_size;
+    AVFrame *delay_frame;
+} CompensationDelayContext;
+
+#define OFFSET(x) offsetof(CompensationDelayContext, x)
+#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption compensationdelay_options[] = {
+    { "mm",   "set mm distance",    OFFSET(distance_mm), AV_OPT_TYPE_INT,    {.i64=0},    0,  10, A },
+    { "cm",   "set cm distance",    OFFSET(distance_cm), AV_OPT_TYPE_INT,    {.i64=0},    0, 100, A },
+    { "m",    "set meter distance", OFFSET(distance_m),  AV_OPT_TYPE_INT,    {.i64=0},    0, 100, A },
+    { "dry",  "set dry amount",     OFFSET(dry),         AV_OPT_TYPE_DOUBLE, {.dbl=0},    0,   1, A },
+    { "wet",  "set wet amount",     OFFSET(wet),         AV_OPT_TYPE_DOUBLE, {.dbl=1},    0,   1, A },
+    { "temp", "set temperature °C", OFFSET(temp),        AV_OPT_TYPE_INT,    {.i64=20}, -50,  50, A },
+    { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(compensationdelay);
+
+// The maximum distance for options
+#define COMP_DELAY_MAX_DISTANCE            (100.0 * 100.0 + 100.0 * 1.0 + 1.0)
+// The actual speed of sound in normal conditions
+#define COMP_DELAY_SOUND_SPEED_KM_H(temp)  1.85325 * (643.95 * pow(((temp + 273.15) / 273.15), 0.5))
+#define COMP_DELAY_SOUND_SPEED_CM_S(temp)  (COMP_DELAY_SOUND_SPEED_KM_H(temp) * (1000.0 * 100.0) /* cm/km */ / (60.0 * 60.0) /* s/h */)
+#define COMP_DELAY_SOUND_FRONT_DELAY(temp) (1.0 / COMP_DELAY_SOUND_SPEED_CM_S(temp))
+// The maximum delay may be reached by this filter
+#define COMP_DELAY_MAX_DELAY               (COMP_DELAY_MAX_DISTANCE * COMP_DELAY_SOUND_FRONT_DELAY(50))
+
+static int query_formats(AVFilterContext *ctx)
+{
+    AVFilterChannelLayouts *layouts;
+    AVFilterFormats *formats;
+    static const enum AVSampleFormat sample_fmts[] = {
+        AV_SAMPLE_FMT_DBLP,
+        AV_SAMPLE_FMT_NONE
+    };
+    int ret;
+
+    layouts = ff_all_channel_counts();
+    if (!layouts)
+        return AVERROR(ENOMEM);
+    ret = ff_set_common_channel_layouts(ctx, layouts);
+    if (ret < 0)
+        return ret;
+
+    formats = ff_make_format_list(sample_fmts);
+    if (!formats)
+        return AVERROR(ENOMEM);
+    ret = ff_set_common_formats(ctx, formats);
+    if (ret < 0)
+        return ret;
+
+    formats = ff_all_samplerates();
+    if (!formats)
+        return AVERROR(ENOMEM);
+    return ff_set_common_samplerates(ctx, formats);
+}
+
+static int config_input(AVFilterLink *inlink)
+{
+    AVFilterContext *ctx = inlink->dst;
+    CompensationDelayContext *s = ctx->priv;
+    unsigned min_size, new_size = 1;
+
+    s->delay = (s->distance_m * 100. + s->distance_cm * 1. + s->distance_mm * .1) *
+               COMP_DELAY_SOUND_FRONT_DELAY(s->temp) * inlink->sample_rate;
+    min_size = inlink->sample_rate * COMP_DELAY_MAX_DELAY;
+
+    while (new_size < min_size)
+        new_size <<= 1;
+
+    s->delay_frame = av_frame_alloc();
+    if (!s->delay_frame)
+        return AVERROR(ENOMEM);
+
+    s->buf_size                    = new_size;
+    s->delay_frame->format         = inlink->format;
+    s->delay_frame->nb_samples     = new_size;
+    s->delay_frame->channel_layout = inlink->channel_layout;
+
+    return av_frame_get_buffer(s->delay_frame, 32);
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+    AVFilterContext *ctx = inlink->dst;
+    CompensationDelayContext *s = ctx->priv;
+    const unsigned b_mask = s->buf_size - 1;
+    const unsigned buf_size = s->buf_size;
+    const unsigned delay = s->delay;
+    const double dry = s->dry;
+    const double wet = s->wet;
+    unsigned r_ptr, w_ptr;
+    AVFrame *out;
+    int n, ch;
+
+    out = ff_get_audio_buffer(inlink, in->nb_samples);
+    if (!out) {
+        av_frame_free(&in);
+        return AVERROR(ENOMEM);
+    }
+    av_frame_copy_props(out, in);
+
+    for (ch = 0; ch < inlink->channels; ch++) {
+        const double *src = (const double *)in->extended_data[ch];
+        double *dst = (double *)out->extended_data[ch];
+        double *buffer = (double *)s->delay_frame->extended_data[ch];
+
+        w_ptr =  s->w_ptr;
+        r_ptr = (w_ptr + buf_size - delay) & b_mask;
+
+        for (n = 0; n < in->nb_samples; n++) {
+            const double sample = src[n];
+
+            buffer[w_ptr] = sample;
+            dst[n] = dry * sample + wet * buffer[r_ptr];
+            w_ptr = (w_ptr + 1) & b_mask;
+            r_ptr = (r_ptr + 1) & b_mask;
+        }
+    }
+    s->w_ptr = w_ptr;
+
+    av_frame_free(&in);
+    return ff_filter_frame(ctx->outputs[0], out);
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+    CompensationDelayContext *s = ctx->priv;
+
+    av_frame_free(&s->delay_frame);
+}
+
+static const AVFilterPad compensationdelay_inputs[] = {
+    {
+        .name         = "default",
+        .type         = AVMEDIA_TYPE_AUDIO,
+        .config_props = config_input,
+        .filter_frame = filter_frame,
+    },
+    { NULL }
+};
+
+static const AVFilterPad compensationdelay_outputs[] = {
+    {
+        .name = "default",
+        .type = AVMEDIA_TYPE_AUDIO,
+    },
+    { NULL }
+};
+
+AVFilter ff_af_compensationdelay = {
+    .name          = "compensationdelay",
+    .description   = NULL_IF_CONFIG_SMALL("Audio Compensation Delay Line."),
+    .query_formats = query_formats,
+    .priv_size     = sizeof(CompensationDelayContext),
+    .priv_class    = &compensationdelay_class,
+    .uninit        = uninit,
+    .inputs        = compensationdelay_inputs,
+    .outputs       = compensationdelay_outputs,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 63b8fdbd05..a3f6e6228a 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -86,6 +86,7 @@ void avfilter_register_all(void)
     REGISTER_FILTER(CHANNELSPLIT,   channelsplit,   af);
     REGISTER_FILTER(CHORUS,         chorus,         af);
     REGISTER_FILTER(COMPAND,        compand,        af);
+    REGISTER_FILTER(COMPENSATIONDELAY, compensationdelay, af);
     REGISTER_FILTER(DCSHIFT,        dcshift,        af);
     REGISTER_FILTER(DYNAUDNORM,     dynaudnorm,     af);
     REGISTER_FILTER(EARWAX,         earwax,         af);
diff --git a/libavfilter/version.h b/libavfilter/version.h
index ed3b642c25..b15cc70437 100644
--- a/libavfilter/version.h
+++ b/libavfilter/version.h
@@ -30,7 +30,7 @@
 #include "libavutil/version.h"
 
 #define LIBAVFILTER_VERSION_MAJOR   6
-#define LIBAVFILTER_VERSION_MINOR  15
+#define LIBAVFILTER_VERSION_MINOR  16
 #define LIBAVFILTER_VERSION_MICRO 100
 
 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \